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Old 26th December 2006, 06:37 AM   #11
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Quote:
Originally posted by EdT
I just need a simple good sounding audio player application, no mixing or ripping required. Some of those proggies are over 50mb for things that I don't need.

Can someone evaluate the soud quality of XMPlay and feedback on the sound quality ? After playing with it for 2 days now I am able to adjust it to hear a larger soundstage and details that I could not hear from Windows Media Player.

Free download here:
http://support.xmplay.com/

Hears an excerpt from the author of XMPlay, can someone make sense of this ?:

XMPlay's got the best (most accurate to FT2) XM reproduction to be heard
from any player. All features/effects (including several FT2 quirks) are
100% supported. The IT reproduction is also practically balls-on, with full
support for all effects/NNA/DCA/filters etc... additional features like
stereo samples and DMO effects are also fully supported.
The last time I looked at XMplay, it wasn't even supporting ASIO or KS! That might have changed. You tell me.
To me tha lack of ASIO or KS support is an absolute NO GO.

In any case the player you have to look at should support full-file-buffering during playback.
Foobar and Samplitude are providing this feature.
If the player is not supporting it running tracks from RAMDISK is the option you'd go for.

You should also be able to play with output-buffer sizes!! This can make huge differeneces depending on your soundcard.
Samplitude provides best features to check this out.

The good thing with all the software players. Install em all! And make up your own mind!

With Linux you won't have the ASIO or KS hassle!
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Old 26th December 2006, 09:31 AM   #12
lykkedk is offline lykkedk  Denmark
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Interesting thread. I am playing off with mplayer(linux) and can't hear any difference in soundquality compared to windows foobar, but maybe it would be the rest of my setup, causing this ?
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Old 26th December 2006, 10:02 AM   #13
Nixie is offline Nixie  Canada
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As a computer scientist, I fail to see how one player program can produce a sound different from the other. The difference can only be where the D/A conversion happens -- on your soundcard. If you use an external DAC, then the interface can have a jitter effect. But this has nothing to do with the software that sends the digital signal to the sound card, which it buffers anyway. The player's job normally doesn't touch the signal content, it's just for choosing which song it streams asynchronously to the sound hardware (which buffers it so any timing effects are not an issue). The only way the player software can make a difference is if you're using DSP plugins. Frankly, this thread is more fit for AudioAsylum among the discussions of shakti stones and cable stands.
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Old 26th December 2006, 11:50 AM   #14
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Quote:
Originally posted by Nixie
As a computer scientist, I fail to see how one player program can produce a sound different from the other. The difference can only be where the D/A conversion happens -- on your soundcard. If you use an external DAC, then the interface can have a jitter effect. But this has nothing to do with the software that sends the digital signal to the sound card, which it buffers anyway. The player's job normally doesn't touch the signal content, it's just for choosing which song it streams asynchronously to the sound hardware (which buffers it so any timing effects are not an issue). The only way the player software can make a difference is if you're using DSP plugins. Frankly, this thread is more fit for AudioAsylum among the discussions of shakti stones and cable stands.
"The audiophile Audio world meets the Computer world!"

Good to have a computer scientist around, who knows about Shakti Stones!

Perhaps having a computer scientist around here might help to get an idea how to get the perfect PCM stream needed for audiophile Audio out of the PC.
In any case one shoudn't be preoccupied and put earlier discussions into the voodoo corner!

The PC is not a very audiophile device, by its nature.
The PC (under Microsoft conditions) is not typically supposed to run realtime applications.
Audiphile PC audio comes down to a realtime, clock perfect, bit-perfect PCM stream.
A typical PC setup is not able to deliver this and would not be able to challenge a well done CD player, that's for sure.

In general problems occur due to interfacing problems, buffer issues, free running asynchrounous clocks, digital noise and so forth.
In general you can't count on a continous bitstream, even if it is
bit-perfect. Yes we're talking of a pure PCM stream, which is forwarded to your soundcard/DAC.
Unfortunately most of the soundcards do convert the PCM stream as it comes in. "S..t in = S..t out"

ASIO and KS wouldn't have been developed if your statement
would be correct!

Under Linux as far as I tested it, the differences are much less obvious than under Windows, perhaps under Vista this will change a bit.
Still - even under Linux you need a proper interface setup, and a quiet, noiseless PC environment.


The good thing, as said before, you and everybody else can
try them all out. Make up your own mind.
The recommended players or tweaks do not cost anything during the demo periods. (compared to a Shakti stone at a price of 100$. )


And as a final word - Believe me there are others around here having a University grade engineering backround!
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Old 26th December 2006, 11:54 AM   #15
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Quote:
Originally posted by Nixie
As a computer scientist, I fail to see how one player program can produce a sound different from the other. ....... Frankly, this thread is more fit for AudioAsylum among the discussions of shakti stones and cable stands.
Unfortunately the voice of reason is rarely listened to when it comes to audio (me? synical? oh yeah!)

There are small variations in the way MP3s are decoded but when it comes to FLAC, WAV and other bit for bit formats there isn't. IF you're not getting bit for bit then its faulty.
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Old 26th December 2006, 10:06 PM   #16
deandob is offline deandob  Australia
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Default Use of a ramdisk sounds better??

I have a couple of PC's tuned for optimum audio response (bypass Kmixer, use of a good PC player, lossless compression, quiet fans, pro-audio soundcards etc) and have to say I dont believe that streaming music from hard disk vs a ramdisk makes a difference. The hard disks have either a 8Mb or 16Mb onboard buffer that streams the music and any temporal domain problems would manifest themselves in pops or clicks not distortion in the analog sense. I did try the ramdisk (Foobar full file buffer) and it made no audio difference as expected.

Other facts about PC audio worth mentioning:
- Windows XP audio mixer (kmixer) will slightly add distortion due to its action. This is mostly bypassed if the system volume is at 100% and even better use kernel streaming or ASIO to bypass.
- Onboard soundcards can sound good but you need the pro-audio cards which take some care with power line filtering, decent onboard clocks etc. EMU 0404 is great value and sounds excellent after modding the output stage, the EMU 1820M is even better, and best still is the Lynx and RME cards (seriously well engineered)
- There is no difference between WAV and lossless codec. The output of a lossless codec is bit perfect with the source so cannot sound different.
- Use a decent player that does not muck with the sound. Foobar2000 or winamp with all the DSP stuff turned off will delivery the bits to the soundcard unmangled.
- Having all your music on hard disk gives you another dimension to your music (search, playlists, quick access etc).

Regards,
Dean
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Old 26th December 2006, 11:39 PM   #17
elg2001 is offline elg2001  United States
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I have to disagree about the winamp comment. I've compared Winamp to Windows Media Player quite heavily, and Windows Media Player sounds better every time. Both players have their equalizers off and program volume at 100%.

I've only compared those two. I haven't looked into Foobar, but VLC sounds pretty darn good too. Winamp is pretty lousy sounding IMO.
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Old 27th December 2006, 12:01 AM   #18
Daveis is offline Daveis  United States
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Has anyone tested the "bit-perfectness" of RME, Lynx, MAudio cards with Windows XP running with and without ASIO?

Winamp with ASIO seems to work well for me as does WMP9 and later.

It seems to me the quality of the supplied sound card drivers has alot to do with sound quality under Windows -- more so than whether you are using WDM or ASIO.

Also, I've had better luck with profesional-type PCI sound cards rather than motherboard AC97, or even firewire based devices.

Can anyone verify whether kmixer comes into play when running Windows Media player and using an RME card?
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Old 27th December 2006, 12:08 AM   #19
deandob is offline deandob  Australia
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I think the RME use a driver that bypasses kmixer (from memory of other posters) and it is about the only card/driver that does. The role of the driver (like the player) is not to muck with the data, but to connect & pass the digital stream to the soundcard DAC.

An easy test for bit perfect is to connect your PC digital out to a AV processor and play a DTS encoded CD or file (not DVD). If there is any bit mangling going on the processor will just output noise. If you hear music you are bit perfect. You can find DTS file samples if you search on the net.

Another good test is to use the "udial" file (in lossless format) which sends a very high frequency & level single tone through your system. If you can hear intermodulation and distortion artifacts (easy with a single tone) then your system is not setup right.

Regards,
Dean
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Old 27th December 2006, 12:28 AM   #20
Nixie is offline Nixie  Canada
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Quote:
Originally posted by soundcheck
ASIO and KS wouldn't have been developed if your statement
would be correct!
I'm not familiar with KS, but the purpose of ASIO is to provide low latency and input-output synchronization -- how would either contribute to sound quality? Indeed, quality is compromised exactly when you have a no latency synchronous interface (i.e. S/PDIF), since the synchronization implies analog timing information.

The sound card is the weakest link. Doing your D/A conversion inside the interference-full PC case is not conductive to good sound. I've been using an outboard DAC for years.

There is no audible difference between playing something on Gigastudio with the 24-bit PMI Emperor grand piano sample set through GSIF interface, and then playing the same recorded as a .wav over DirectSound. I would expect both end up having streams buffered enough in the sound card before being sent out on the S/PDIF cable that without any DSP going on, there would be no difference. While I can imagine that changing the driver and soundcard firmware can make a difference, to think that just changing the player would do so is ludicrous.

Disclaimer: I'm using an ASRC in my DAC so if by any chance there is a limited difference in jitter, it would not be noticeable due to jitter attenuation by the ASRC (though I put my bet on no difference whatsoever on the digital stream).
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